KR20210145832A - Optimizing loudness and dynamic range across different playback devices - Google Patents

Abstract

Embodiments provide a method and system for receiving metadata associated with audio data in a bitstream and analyzing the metadata to determine whether a loudness parameter for a first group of audio reproduction devices is available in the bitstream is about In response to determining that the parameters are present for the first group, the system uses the parameters and audio data to render audio. In response to determining that the loudness parameters do not exist for the first group, the system analyzes one or more characteristics of the first group and determines the parameter based on the one or more characteristics.

Description

Optimizing loudness and dynamic range across different playback devices {OPTIMIZING LOUDNESS AND DYNAMIC RANGE ACROSS DIFFERENT PLAYBACK DEVICES}

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is incorporated herein by reference in its entirety, in U.S. Provisional Application Serial Nos. 61/754,882, filed on January 21, 2013; U.S. Provisional Application No. 61/809,250, filed April 5, 2013; and U.S. Provisional Application No. 61/824,010, filed on May 16, 2013.

One or more embodiments relate generally to audio signal processing, and more particularly to processing audio data bitstreams having metadata indicative of loudness and dynamic range characteristics of audio content based on playback environments and devices. it's about

The subject matter discussed in the background section should not be presumed to be prior art merely because it is recited in the background section. Similarly, problems mentioned in the background section or associated with the subject matter of the background section should not be presumed to have been previously recognized in the prior art. The subject matter in the background part merely represents different approaches, which in nature and per se can also be inventions.

The dynamic range of an audio signal is the ratio between the maximum and minimum possible values of the sound contained in the signal, and is usually measured as a decibel (base-10) value. In many audio processing systems, dynamic range control (or dynamic range compression, DRC) provides large sound to fit wide dynamic range source content into a narrower recorded dynamic range that can be more easily stored and played back using electronic equipment. used to reduce the level of noise and/or amplify the level of quiet sounds. For audio/visual (AV) content, a dialog reference level may be used to define a “null” point for compression via the DRC mechanism. The DRC operates to boost content below the dialog reference level and cut the content above the reference level.

In known audio encoding systems, metadata associated with an audio signal is used to set the DRC level based on the type of content and its intended use. DRC mode sets the amount of compression applied to the audio signal and defines the output reference level of the decoder. These systems are programmed with the encoder and can be limited to two DRC level settings selected by the user. For example, a Dialnorm (dialog normalization) value of -31 dB (line) is traditionally used for content played on AVR or full dynamic range capable devices, and a Dialnorm value of -20 dB (RF) is used for television sets. or for content played on similar devices. This type of system allows a single audio bitstream to be used in two general but very different playback scenarios through the use of two different sets of DRC metadata. However, these systems are limited to preset Dialnorm values and have not been optimized for playback in a wide variety of different playback devices and listening environments currently available through the advent of digital media and Internet-based streaming technology.

In current metadata-based audio encoding systems, a stream of audio data may include both audio content (eg, one or more channels of audio content) and metadata representing at least one characteristic of the audio content. . For example, in the AC-3 bitstream, there are several audio metadata parameters specifically intended for use when changing the sound of a program delivered to the listening environment. One of the metadata parameters is a Dialnorm parameter indicating an average loudness level (or average loudness of content) of a dialog generated in an audio program, and is used to determine an audio reproduction signal level.

During playback of a bitstream containing a sequence of different audio program segments (each having a different Dialnorm parameter), the AC-3 decoder changes the playback level or loudness of the segment so that the perceived loudness of the segment's dialog is at a consistent level. Use the Dialnorm parameter of each segment to perform some type of loudness processing. Each encoded audio segment (item) in the sequence of encoded audio items has (generally) a different Dialnorm parameter and may require application of different gain amounts to different levels of items during playback, but the decoder Each level of the items will be scaled so that the loudness or reproduction level of the dialog for the item is the same or very similar.

In some embodiments, the Dialnorm parameter is set by the user, and there is a default Dialnorm value if no value is set by the user, but is not automatically generated. For example, the content generator may make loudness measurements with a device external to the AC-3 encoder and then send the result (representing the loudness of the audio program's voice dialog) to the encoder to set the Dialnorm value to the Dialnorm value. to set the value. Therefore, you have to rely on the content generator to set the Dialnorm parameter correctly.

There are several different reasons why the Dialnorm parameter in an AC-3 bitstream may be incorrect. First, each AC-3 encoder has a default Dialnorm value used during generation of the bitstream unless the Dialnorm value is set by the content generator. This default value can be significantly different from the actual dialog loudness level of the audio. Second, even if the content generator measures the loudness and sets the Dialnorm value accordingly, a loudness measurement algorithm or meter that does not follow the recommended loudness measurement method may be used, which may cause an inaccurate Dialnorm value. Third, even if the AC-3 bitstream is generated with the measured Dialnorm value and set correctly by the content generator, it may have been changed to an incorrect value by an intermediate module during transmission and/or storage of the bitstream. For example, in television broadcast applications it is not uncommon for AC-3 bitstreams to be decoded, modified and then re-encoded using incorrect Dialnorm metadata information. Therefore, the Dialnorm value included in the AC-3 bitstream may be inaccurate or erroneous and may therefore have a negative impact on the quality of the listening experience.

Furthermore, the Dialnorm parameter does not indicate the loudness processing state of the corresponding audio data (eg, what type(s) of loudness processing has been performed on the audio data). Additionally, currently deployed loudness and DRC systems, such as those in Dolby Digital (DD) and Dolby Digital Plus (DD+) systems, are designed to render AV content in a consumer's living room or movie theater. In order to adapt such content to playback in other environments and listening equipment (eg mobile device), post-processing must be 'blindly' applied at the playback device to fit AV content to the listening environment. In other words, the post-processor (or decoder) assumes that the loudness level of the received content is at a certain level (eg -31 or -20 dB) and the post-processor returns to a predetermined fixed target level suitable for the particular device. Set the level. If the estimated loudness level or the predetermined target level is incorrect, post-processing may have an effect opposite to its intended effect; That is, post-processing may make the output audio less desirable to the user.

Although the disclosed embodiments are not limited for use with AC-3 bitstreams, E-AC-3 bitstreams, or Dolby E bitstreams, for convenience, these bitstreams are will be discussed together. Dolby, Dolby Digital, Dolby Digital Plus, and Dolby E are trademarks of Dolby Laboratories Licensing Corporation. Dolby Laboratories offers proprietary implementations of AC-3 and E-AC-3, known as Dolby Digital and Dolby Digital Plus, respectively.

The present invention provides an improved system and method for optimizing loudness and dynamic range across different playback devices.

Embodiments decode the audio data by receiving a bitstream comprising metadata associated with the audio data, wherein a loudness parameter for a first group of audio reproduction devices is available in the bitstream. A method for analyzing metadata in a bitstream. In response to determining that the parameters for the first group exist, a processing component uses the parameters and audio data to render audio. In response to determining that the loudness parameters for the first group do not exist, the processing component analyzes one or more characteristics of the first group and determines the parameter based on the one or more characteristics. The method may further use the parameters and audio data to render audio by sending the parameters and audio data to a downstream module that renders the audio for playback. The parameters and audio data may also be used to render audio by rendering the audio data based on the parameters and audio data.

In an embodiment, the method also comprises determining an output device to render the received audio stream, and determining whether the output device belongs to the first group of audio reproduction devices, wherein parsing the metadata in the stream to determine whether a loudness parameter is available for the first group of audio reproduction devices in: determining that the output device belongs to the first group of audio reproduction devices is executed after the step. In an embodiment, determining that the output device belongs to the first group of audio reproduction devices comprises: an indication indicating the identity of the output device or indicating the identity of a group of devices comprising the output device receiving from a module connected to the output device, and determining that the output device belongs to the first group of audio reproduction devices based on the received indication.

Embodiments also relate to an apparatus or system comprising processing components for performing the operations described in the encoding method embodiments above.

Embodiments also decode the audio data by receiving audio data and metadata associated with the audio data, and determine whether loudness information associated with loudness parameters for a first group of audio devices is available in the stream. parses metadata in the bitstream to determine loudness information from the stream, in response to a determination that loudness information for the first group exists, determines loudness information from the stream, the audio data for use in rendering audio; A method of transmitting loudness information or, if no loudness information exists for the first group, determining loudness information associated with an output profile, and transmitting the determined loudness information for the output profile for use in rendering audio. is about In one embodiment, determining loudness information associated with an output profile comprises analyzing characteristics of the output profile, determining the parameters based on the characteristics, and transmitting the determined loudness information comprises: and transmitting the determined parameters. The loudness information may include loudness parameters for the output profile or characteristics thereof. In an embodiment, the method may further comprise determining a low bit rate encoded stream to be transmitted, wherein the loudness information includes characteristics for one or more output profiles.

Embodiments also relate to an apparatus or system comprising processing components for performing the operations described in the decoding method embodiments above.

The system according to the invention generates appropriate gains based on loudness control and dynamic range requirements at the encoder or at the decoder, under control from the encoder through parameterization of the original gains to reduce the data rate, , also allows other metadata (internal or external) parameters to be used to properly control loudness and dynamic range gains and/or profiles.

In the following drawings, like reference numbers are used to denote like elements. Although the following figures depict various examples, the implementations described herein are not limited to the examples depicted in the figures.
1 is a block diagram of an embodiment of an audio processing system configured to perform optimization of loudness and dynamic range, in some embodiments;
FIG. 2 is a block diagram of an encoder for use in the system of FIG. 1 , in some embodiments;
3 is a block diagram of a decoder for use in the system of FIG. 1 , in some embodiments;
Fig. 4 is a diagram of an AC-3 frame divided into segments;
5 is a diagram of a synchronization information (SI) segment of an AC-3 frame divided into segments;
6 is a diagram of a bitstream information (BSI) segment of an AC-3 frame divided into segments;
Fig. 7 is a diagram of an E-AC-3 frame divided into segments;
8 is a table illustrating the format of specific frames and metadata of an encoded bitstream, in some embodiments.
9 is a table illustrating the format of loudness processing state metadata, in some embodiments.
FIG. 10 is a more detailed block diagram of the audio processing system of FIG. 1 that may be configured to perform optimization of loudness and dynamic range, in some embodiments;
11 is a table illustrating different dynamic range requirements for various playback devices and background listening environments in an example use case;
12 is a block diagram of a dynamic range optimization system, in an embodiment;
13 is a block diagram illustrating the interface between different profiles for various different playback device classes, in an embodiment;
14 is a table illustrating the correlation between long term loudness and short term dynamic range for a plurality of defined profiles, in an embodiment.
Fig. 15 illustrates examples of loudness profiles for different types of audio content, in an embodiment;
16 is a flow diagram illustrating a method of optimizing loudness and dynamic range across playback devices and applications, in an embodiment.

Definitions and nomenclature

Throughout this disclosure, including in the claims, representations that perform an operation "on" a signal or data (eg, filter, scale, transform, or apply a gain to the signal or data) are It is used in a broad sense to refer to performing an operation directly on, or on a processed version of a signal or data (eg, on a version of the signal that has been pre-filtered or pre-processed prior to performance of the operation). The expression (“system”) is used in its broadest sense to refer to a device, system, or subsystem. For example, a subsystem implementing a decoder may be referred to as a decoder system, and a system including such a subsystem (eg, a system that generates X output signals in response to multiple inputs, wherein the subsystem is M ) and other XM inputs are received from an external source) can also be referred to as a decoder system. The term (“processor”) refers to a system or device programmable or otherwise configurable (eg, in software or firmware) to perform operations on data (eg, audio, or video or other image data). used in a broad sense for Examples of processors include a field-programmable gate array (or other configurable integrated circuit or chipset), a digital signal processor programmed and/or otherwise configured to perform pipeline processing on audio or other sound data, a programmable general-purpose processors or computers, and programmable microprocessor chips or chipsets.

The expressions "audio processor" and "audio processing unit" are used interchangeably, and in a broad sense are used to refer to a system configured to process audio data. Examples of audio processing units include, but are not limited to, encoders (eg, transcoders), decoders, codecs, pre-processing systems, post-processing systems, and bitstream processing systems (sometimes bit stream processing tools). The expression “processing state metadata” (eg, the expression “loudness processing state metadata”) represents distinct and different data from the corresponding audio data (the audio content of the audio data stream that also includes the processing state metadata). The processing state metadata is associated with the audio data and indicates a loudness processing state of the corresponding audio data (eg, what type(s) of processing has already been performed on the audio data), and optionally also at least one of the audio data. characteristics or characteristics of In some embodiments, the association of audio data with processing state metadata is time-synchronous. Thus, the current (most recently received or updated) processing state metadata indicates that the corresponding audio data simultaneously contains the results of audio data processing of the indicated type(s). In some cases, processing state metadata may include some or all of the processing history and/or parameters used in and/or obtained from the indicated types of processing. Additionally, the processing state metadata may include at least one characteristic or characteristic of the corresponding audio data that has been calculated from or extracted from the audio data. The processing state metadata may also include other metadata not related to or derived from any processing of the corresponding audio data. For example, third party data, tracking information, identifiers, proprietary or standard information, user annotation data, user preference data, etc. may be added by a particular audio processing unit to communicate to other audio processing units. have.

The expression "loudness processing state metadata" (or "LPSM") indicates and optionally corresponds to a loudness processing state (eg, what type(s) of loudness processing has been performed on the audio data) of the corresponding audio data. Indicate processing state metadata representative of at least one characteristic or characteristic (eg, loudness) of the audio data. Loudness processing state metadata may include data (eg, other metadata) that is not loudness processing state metadata (ie, when considered alone). The terms “couple” or “coupled” are used to mean either direct or indirect connection.

Systems and methods are described for an audio encoder/decoder that non-destructively normalizes the loudness and dynamic range of audio across various devices that require or use different target loudness values and have different dynamic range capabilities. Methods and functional components according to some embodiments transmit information about audio content from an encoder to a decoder for one or more device profiles. A device profile specifies a desired target loudness and dynamic range for one or more devices. The system is extensible, so new device profiles with different “nominal” loudness targets can be supported.

In an embodiment, the system generates appropriate gains based on loudness control and dynamic range requirements at the encoder or at the decoder, under control from the encoder via parameterization of the original gains to reduce the data rate. generate The dynamic range system includes two mechanisms for implementing loudness control: an artistic dynamic range profile that provides content creator control over how audio is played, and ensuring that overloading does not occur for various playback profiles. separate protective mechanisms for The system is also configured such that other metadata (internal or external) parameters can be used to properly control loudness and dynamic range gains and/or profiles. The decoder is configured to support an n-channel auxiliary input to leverage decoder-side loudness and dynamic range settings/processing.

In some embodiments, the loudness processing state metadata (LPSM) also includes one or more reserved fields (or slots) of metadata segments of an audio bitstream that contain audio data in other segments (audio data segments). ) is built into For example, at least one segment of each frame of the bitstream comprises an LPSM, and at least one other segment of the frame comprises corresponding audio data (ie, audio whose loudness processing state and loudness is represented by the LPSM). data) are included. In some embodiments, the data volume of the LPSM may be small enough to be carried without affecting the bit rate allocated to carry the audio data.

Passing loudness processing state metadata in an audio data processing chain is particularly useful when you require two or more audio processing units to operate in concert with each other throughout the processing chain (or content lifecycle). If the audio bitstream does not include loudness processing state metadata, media processing problems such as quality, level and spatial degradation may occur, for example, when more than one audio codec is used in a chain and single-stage volume leveling is not applied to the media consuming device (or It may occur when applied more than once during the journey of the bitstream for a rendering point of the audio content of the bitstream).

Loudness and dynamic range metadata processing system

1 is a block diagram of an embodiment of an audio processing system that may be configured to perform optimization of loudness and dynamic range, in some embodiments using certain metadata processing (eg, pre-processing and post-processing) components; am. 1 illustrates an exemplary audio processing chain (audio data processing system), wherein one or more of the elements of the system may be configured in accordance with an embodiment of the present invention. The system 10 of FIG. 1 includes the following elements, coupled together as shown: a pre-processing unit 12 , an encoder 14 , a signal analysis and metadata correction unit 16 , a transcoder ( 18 ), decoder 20 , and post-processing unit 24 . As a variation on the illustrated system, one or more of the elements are omitted, or additional audio data processing units are included. For example, in one embodiment, the post-processing unit 22 is part of the decoder 20 instead of a separate unit.

In some implementations, the pre-processing unit of FIG. 1 is configured to accept PCM (time-domain) samples including audio content as input 11 , and output the processed PCM samples. The encoder 14 may be configured to accept the PCM samples as input and output an encoded (eg, compressed) audio bitstream representing the audio content. The data of the bitstream representing the audio content is sometimes referred to herein as "audio data". In one embodiment, the audio bitstream output from the encoder includes audio data as well as loudness processing state metadata (and optionally also other metadata).

The signal analysis and metadata correction unit 16 accepts one or more encoded audio bitstreams as input and performs signal analysis, thereby determining (eg, verifying) that the processing state metadata in each encoded audio bitstream is correct. )can do. In some embodiments, the verification may be performed by a state verifier component, such as element 102 shown in FIG. 2 , one such verification technique is described below in the context of state verifier 102 . In some embodiments, unit 16 is included in the encoder and the verification is done by unit 16 or verifier 102 . If the signal analysis and metadata correction unit finds that the metadata included is invalid, the metadata correction unit 16 performs signal analysis to determine the correct value(s) and recalls the incorrect value(s). Replace with the exact value(s) determined. Accordingly, each encoded audio bitstream output from the signal analysis and metadata correction unit may include not only the encoded audio data but also the corrected processing state metadata. The signal analysis and metadata correction unit 16 may be part of the pre-processing unit 12 , the encoder 14 , the transcoder 18 , the decoder 20 , or the post-processing unit 22 . Alternatively, the signal analysis and metadata correction unit 16 may be a separate unit or part of another unit in the audio processing chain.

Transcoder 18 accepts encoded audio bitstreams as input and in response (eg, by decoding the input stream and re-encoding the decoded stream into a different encoding format) modified (eg, differently encoded) ) can output audio bitstreams. The audio bitstream output from the transcoder includes encoded audio data as well as loudness processing state metadata (and optionally also other metadata). The metadata may be included in the bitstream.

The decoder 20 of FIG. 1 may accept as input encoded (eg, compressed) audio bitstreams and (in response) output streams of decoded PCM audio samples. In one embodiment, the output of the decoder is or comprises any of the following: a stream of audio samples, and a corresponding stream of loudness processing state metadata (and optionally also other metadata) extracted from the input encoded bitstream. ; a stream of audio samples and a corresponding stream of control bits determined from loudness processing state metadata (and optionally also other metadata) extracted from the input encoded bitstream; or a corresponding stream of processing state metadata or a stream of audio samples without control bits determined from processing state metadata. In this last case, the decoder extracts the loudness processing state metadata (and/or other metadata) from the input encoded bitstream and extracts the extracted metadata, even if it does not output the extracted metadata or the control bits determined therefrom. At least one operation (eg, verification) may be performed.

By configuring the post-processing unit of Fig. 1 according to an embodiment of the present invention, the post-processing unit 22 accepts a stream of decoded PCM audio samples, and receives the received loudness processing state metadata (and optional also other metadata), or post-processing on it (e.g. audio content) using control bits received with the samples (determined by the decoder from loudness processing state metadata and optionally also other metadata) volume leveling). The post-processing unit 22 is optionally also configured to render the post-processed audio content for playback by one or more speakers. These speakers may be implemented in any of a variety of different listening devices or items of playback equipment, such as computers, televisions, stereo systems (home or cinema), mobile phones, and other portable playback devices. . The speakers may be of any suitable size and power rating, and may be provided in the form of stand-alone drivers, speaker enclosures, surround-sound systems, soundbars, headphones, earbuds, and the like.

Some embodiments provide that audio processing units (eg, encoders, decoders, transcoders, and pre- and post-processing units) are indicated by loudness processing state metadata received by the audio processing units, respectively. As described above, it provides an enhanced audio processing chain that adapts their respective processing to be applied to audio data according to the co-occurrence state of the media data. Audio data input 11 to any audio processing unit (eg, the encoder or transcoder in FIG. 1 ) of the system 100 may include audio data (eg, encoded audio data) as well as loudness processing state metadata ( and optionally also other metadata). Such metadata may be included in the input audio by another element or another source in accordance with some embodiments. the processing unit receiving (with metadata) the input audio performs at least one action (eg, validation) on or in response to the metadata (eg, adaptive processing of the input audio); It may also optionally be configured to include in its output audio metadata, a processed version of the metadata, or control bits determined from the metadata.

An embodiment of the audio processing unit (or audio processor) is configured to perform adaptive processing of the audio data based on the state of the audio data as indicated by the loudness processing state metadata corresponding to the audio data. In some embodiments, if the metadata indicates that loudness processing or similar processing has not yet been performed on the audio data, the adaptive processing is (or includes) loudness processing, but the metadata indicates that such loudness processing or its If it indicates that processing similar to that has already been performed on the audio data, it is not (but does not include) loudness processing. In some embodiments, the adaptive processing performs metadata validation ( eg, performed in the metadata validation sub-unit) or include. In some embodiments, the verification determines the reliability of loudness processing state metadata associated with (eg, included in the bitstream with) audio data. For example, if the metadata is verified to be reliable, results from the previously performed type of audio processing may be reused and additional performance of the same type of audio processing may be avoided. On the other hand, if it is found that the metadata has been altered (or otherwise unreliable), the type of media processing previously known to have been performed (as indicated by the unreliable metadata) is repeated by the audio processing unit. and/or other processing may be performed by the audio processing unit on the metadata and/or audio data. The audio processing unit also determines the loudness processing state (eg, present in the media bitstream) if the unit determines that the processing state metadata is valid (eg, based on a match of the extracted cipher value and the reference cipher value). The metadata may be configured to signal other audio processing units down in the valid enhanced media processing chain.

For the embodiment of FIG. 1 , the pre-processing component 12 may be part of the encoder 14 and the post-processing component 22 may be part of the decoder 22 . Alternatively, the pre-processing component 12 may be implemented as a functional component separate from the encoder 14 . Similarly, post-processing component 22 may be implemented as a functional component separate from decoder 20 .

2 is a block diagram of an encoder 100 that may be used with the system 10 of FIG. 1 . Any of the components or elements of encoder 100 may be hardware, software, or a combination of hardware and software, with one or more processors and/or one or more circuits (eg, ASICs, FPGAs, or other integrated circuits). circuits) can be implemented. The encoder 100 includes a frame buffer 110 , a parser 111 , a decoder 101 , an audio state verifier 102 , a loudness processing stage 103 , an audio stream selection stage 104 , connected as shown. It includes an encoder 105 , a stuffer/formatter stage 107 , a metadata generation stage 106 , a dialog loudness measurement subsystem 108 , and a frame buffer 109 . Optionally, the encoder 100 also includes other processing elements (not shown). The encoder 100 (which is a transcoder) includes in the input bitstream an input audio bitstream (which may be, for example, one of an AC-3 bitstream, an E-AC-3 bitstream, or a Dolby E bitstream). An encoded output audio bitstream (e.g., an AC-3 bitstream, an E-AC-3 bitstream, or a Dolby E bit may be another of the streams). For example, the encoder 100 may output an AC-3 or E-AC- in an input Dolby E bitstream (a format not typically used in consumer devices receiving broadcast audio programs, but used in production and broadcast facilities). 3 format encoded output audio bitstream (suitable for broadcasting to consumer devices).

The system of FIG. 2 also includes an encoded audio delivery subsystem 150 (which stores and/or delivers the encoded bitstreams output from the encoder 100 ) and a decoder 152 . The encoded audio bitstream output from encoder 100 may be stored by subsystem 150 (eg, in the form of a DVD or Blu-ray Disc), or may be stored by subsystem 150 (which may implement a transmit link or network). It may be transmitted by system 150 , or both stored and transmitted by subsystem 150 . The decoder 152 is configured to decode an encoded audio bitstream it receives via the subsystem 150 (generated by the encoder 100 ), and from each frame of the bitstream loudness processing state metadata (LPSM). extracting and generating decoded audio data. In one embodiment, the decoder 152 is pre-configured to perform adaptive loudness processing on audio data decoded using LPSM and/or to perform adaptive loudness processing on audio data decoded using LPSM and forward the decoded audio data and LPSM to a processor. Optionally, decoder 152 includes a buffer, which stores (eg, in a non-transitory manner) an encoded audio bitstream received from subsystem 150 .

Various implementations of encoder 100 and decoder 152 are configured to perform the different embodiments described herein. Frame buffer 110 is a buffer memory coupled to receive an encoded input audio bitstream. In operation, the buffer 110 stores (eg, in a non-transitory manner) at least one frame of the encoded audio bitstream, and the sequence of frames of the encoded audio bitstream is retrieved from the buffer 110 by the parser 110 ) is asserted. Parser 111 extracts loudness processing state metadata (LPSM) and other metadata from each frame of encoded input audio, and converts at least the LPSM to audio state verifier 102, loudness processing stage 103, stage ( 106 ) and subsystem 108 , extracting audio data from the encoded input audio, and asserting the audio data to the decoder 101 . The decoder 101 of the encoder 100 decodes the audio data to generate decoded audio data, and sends the decoded audio data to the loudness processing stage 103, the audio stream selection stage 104, the subsystem 108, and optionally also assert to the state verifier 102 .

The state verifier 102 is configured to authenticate and verify the asserted LPSM (and optionally other metadata). In some embodiments, the LPSM is (or included in) a data block included in an input bitstream (eg, according to an embodiment of the invention). The block is a cryptographic hash (hash-based message authentication code or “HMAC”) for processing the LPSM (and optionally also other metadata) and/or basic audio (provided from decoder 101 to verifier 102 ). It may contain data. The data block may be digitally signed in these embodiments, so that the downstream audio processing unit can authenticate and verify processing state metadata with relative ease.

For example, HMAC is used to generate a digest, and the protection value(s) included in the bitstream of the present invention may include the digest. The digest may be generated for an AC-3 frame as follows: (1) After AC-3 data and LPSM are encoded, frame data bytes (concatenated frame_data#1 and frame_data#2) and LPSM data bytes are used as input to the hashing-function HMAC. Any other data that may be present inside the ancillary data field is not taken into account to calculate the digest. These other data may be bytes that do not belong to either AC-3 data or LPSM data. The guard bits included in the LPSM may not be considered to calculate the HMAC digest. (2) After the digest is calculated, it is written into a bitstream in a field reserved for guard bits. (3) The final step in the generation of a complete AC-3 frame is the generation of a CRC-check. It is written at the very end of the frame and including the LPSM bits, all data belonging to this frame is considered to contain the LPSM bits.

Validation of the LPSM (e.g., at verifier 102) to ensure secure transmission and reception of the LPSM and/or underlying audio data by other cryptographic methods, including, but not limited to, any of one or more non-HMAC cryptographic methods. can be used for For example, verification (using these cryptographic methods) indicates that the loudness processing state metadata contained in the bitstream and the corresponding audio data contained in the bitstream have been subjected to specific loudness processing (as indicated by the metadata); may be performed at each audio processing unit receiving an emblem of the audio bitstream to determine whether (and/or is a result thereof) has not changed after performing this particular loudness processing.

The state verifier 102 asserts the control data to the audio stream selection stage 104 , the metadata generator 106 , and the dialog loudness measurement subsystem 108 to represent the results of the verify operation. In response to the control data, the stage 104 may select (and pass to the encoder 105) one of the following: (1) an adaptively processed output of the loudness processing stage 103 (eg, , when the LPSM indicates that the audio data output from the decoder 101 is not subject to a particular type of loudness processing, and the control bits from the verifier 102 indicate that the LPSM is valid); or (2) the audio data output from the decoder 101 (eg, the LPSM indicates that the audio data output from the decoder 101 has already been subjected to a specific type of loudness processing to be performed by the stage 103 ) and the verifier 102 ) when the control bits from ) indicate that the LPSM is valid). In an embodiment, the loudness processing stage 103 corrects the loudness to a specific target and loudness range.

The stage 103 of the encoder 100 performs adaptive loudness processing on the decoded audio data output from the decoder 101 based on one or more audio data characteristics indicated by the LPSM extracted by the decoder 101 . is configured to perform Stage 103 may be an adaptive transform-domain real-time loudness and dynamic range control processor. Stage 103 may include user input (eg, user target loudness/dynamic range values or Dialnorm values), or other metadata input (eg, one or more types of third-party data, tracking information, identifiers, proprietary or standard information, user annotation data, user preference data, etc.), and/or other inputs (eg, from a fingerprinting process) and use these inputs to process decoded audio data output from decoder 101 . can

The dialog loudness measurement subsystem 108 may, for example, evaluate the LPSM (and/or other metadata) extracted by the decoder 101 when the control bits from the verifier 102 indicate that the LPSM is not valid. used to determine the loudness of segments of decoded audio (from decoder 101 ) representing dialog (or other speech). The operation of the dialog loudness measurement subsystem 108 causes the LPSM to transfer dialog (or other speech) segments of the decoded audio (from the decoder 101) when the control bits from the verifier 102 indicate that the LPSM is valid. It may be disabled when indicating the determined loudness.

Useful tools (eg, Dolby LM100 Loudness Meter) exist to conveniently and easily measure the level of dialogue in audio content. Some embodiments of an APU (eg, stage 108 of encoder 100 ) have an audio bitstream (eg, decoded AC-3 asserted from decoder 101 of encoder 100 to stage 108 ) It is implemented to include (or perform its function) such a tool for measuring the average dialog loudness of the audio content of a bitstream). If stage 108 is performed to measure the actual average dialog loudness of the audio data, the measurement may include separating segments of audio content that usually contain speech. Audio segments, usually speech, are then processed according to a loudness measurement algorithm. For audio data decoded from an AC-3 bitstream, this algorithm may be a standard K-weighted loudness measure (according to international standard ITU-R BS.1770). Alternatively, other loudness measures may be used (eg, those based on psychoacoustic models of loudness).

Separation of speech segments is not necessary to measure the average dialog loudness of the audio data. However, it improves the accuracy of the measurement and provides more satisfactory results from the listener's point of view. Since not all audio content contains dialogue (speech), if there is speech, a measure of the loudness of the entire audio content can provide a sufficient approximation of the dialogue level of the audio.

The metadata generator 106 generates metadata to be included by the stage 107 in the encoded bitstream to be output from the encoder 100 . The metadata generator 106 passes the LPSM (and/or other metadata) extracted by the encoder 101 to the stage 107 (eg, the control bits from the verifier 102 are create a new LPSM (and/or other metadata) or assert the new metadata to stage 107 (e.g., control bits from verifier 102) or when other metadata indicates that it is valid assert the combination of the newly generated metadata and the metadata extracted by the decoder 1010 to the stage 107) or to stage 107 The metadata generator 106 may include the loudness data generated by the subsystem 108 and at least one value indicating the type of loudness processing performed by the subsystem 108, the LPSM , assert to stage 107 for inclusion in the encoded bitstream to be output from encoder 100. Metadata generator 106 includes an LPSM (and optionally also other metadata) to be included in the encoded bitstream and / or may consist of or contain protection bits (hash-based message authentication code or "HMAC") useful for at least one of decryption, authentication, or verification of the underlying audio data to be included in the encoded bitstream ) can be created. The metadata generator 106 may provide these guard bits to the stage 107 for inclusion in the encoded bitstream.

In one embodiment, dialog loudness measurement subsystem 108 processes audio data output from decoder 101 and in response calculates loudness values (eg, gated and non-gated dialog loudness values) and dynamic range values. create In response to these values, the metadata generator 106 generates loudness processing state metadata (LPSM) for inclusion in the encoded bitstream to be output from the encoder 100 (by the stuffer/formatter 107 ). can In an embodiment, the loudness may be calculated based on techniques specified by the ITU-R BS.1770-1 and ITU-R BS.1770-2 standards, or other similar loudness measurement standards. The gated loudness may be dialog-gated loudness or relative-gated loudness, or a combination of these gating loudness types, and the system may use appropriate gating blocks depending on application requirements and system constraints.

Additionally, optionally, or alternatively, subsystems 106 and/or 108 of encoder 100 represent at least one characteristic of audio data for inclusion in an encoded bitstream to be output from stage 107 . Additional analysis of the audio data may be performed to generate metadata. The encoder 105 encodes the audio data output from the selection stage 104 (eg, by performing compression thereon), and stages the encoded audio for inclusion in the encoded bitstream to be output from the stage 107 . assert with (107).

Stage 107 multiplexes the encoded audio from encoder 105 and metadata from generator 106 (including LPSM) to produce an encoded bitstream to be output from stage 107, and thus encoded The bitstream has a format as specified by the embodiment. The frame buffer 109 is a buffer memory that stores (eg, in a non-transitory manner) at least one frame of an encoded audio bitstream output from the stage 107 , wherein the sequence of frames of the encoded audio bitstream includes: It is then asserted from the buffer 109 to the delivery system 150 as output from the encoder 100 .

The LPSM generated by the metadata generator 106 and included in the bitstream encoded by the stage 107 indicates the loudness processing state of the corresponding audio data (eg, what type(s) of loudness processing is performed for the audio data). performed) and the loudness of the corresponding audio data (eg, measured dialog loudness, gated and/or non-gated loudness, and/or dynamic range). Here, “gating” of loudness and/or level measurements performed on audio data means that the calculated value(s) exceeding the threshold are included in the final measurement, either at a loudness threshold or at a specific level (e.g., at the final measurement values - Ignoring short-term loudness values below 60 dBFS). Gating on absolute values indicates a fixed level or loudness, while gating on relative values indicates values that depend on the current “non-gated” measurement.

In some implementations of encoder 100 , the encoded bitstream buffered in memory 109 (and output to delivery system 150 ) is an AC-3 bitstream or an E-AC-3 bitstream, and an audio data segment (eg, segments AB0 through AB5 of the frame shown in FIG. 4 ) and metadata segments, wherein the audio data segments represent audio data, wherein each of at least some of the metadata segments is a loudness processing state meta data (LPSM). Stage 107 inserts the LPSM into the bitstream in the following format. Each of the metadata segments containing the LPSM is stored in the "addbsi" field of the bitstream information ("BSI") segment of the frame of the bitstream, or in the ancillary data field (e.g., shown in FIG. 4) at the end of the frame of the bitstream. included in the AUX segment).

A frame of a bitstream contains one or two metadata segments, each containing an LPSM, and if a frame contains two metadata segments, one is in the addbsi field of the frame and the other is the AUX of the frame. exists in the field. Each metadata segment comprising an LPSM includes an LPSM payload (or container) segment with the following format: a header (eg, a syncword identifying the start of the LPSM payload, followed by at least one identifying value, eg LPSM format version, length, duration, count, and substream associated values shown in Table 2 below); and after the header, at least one dialog display value (eg, the parameter “dialog channel(s)” of Table 2) indicating whether the corresponding audio data displays a dialog or does not display a dialog (eg, a corresponding which channels of audio data display the dialog); at least one loudness rule compliance value indicating whether the corresponding audio data conforms to the indicated set of loudness rules (eg, the parameter “loudness rule type” in Table 2); One or more of at least one loudness processing value indicating at least one type of loudness processing performed on the corresponding audio data (eg, parameters “Dialog Gating Loudness Correction Flag”, “Loudness Correction Type” in Table 2) ); and at least one loudness value indicative of at least one loudness (eg, peak or average loudness) characteristic of the corresponding audio data (eg, parameters “ITU Relative Gating Loudness”, “ITU Speech Gating Loudness” in Table 2) , "ITU (EBU 3341) short-term 3s loudness", and one or more of "real peaks").

In some implementations, each of the metadata segments inserted by stage 107 into an "addbsi" field or ancillary data field of a frame of the bitstream has the following format: a core header (eg, start of a metadata segment) a syncword that identifies , followed by identification values, such as core element version, length, and duration, extended element count, and substream association values as shown in Table 1 below); and, after the core header, at least one protection value useful for at least one of decryption, authentication, or verification of at least one of loudness processing state metadata or corresponding audio data (eg, the HMAC digest and audio fingerprint values of Table 1). ); and LPSM Payload Identification (“ID”) and LPSM Payload Size values that also identify subsequent metadata as an LPSM payload and indicate the size of the LPSM payload, if the metadata segment contains LPSM after the core header.

An LPSM payload (or container) segment (eg, having the above-specified format) conforms to the LPSM Payload ID and LPSM Payload Size values.

In some embodiments, each of the metadata segments in the ancillary data field (or "addbsi" field) of the frame has three levels of structure: a flag indicating whether the ancillary data (or addbsi) field contains metadata , at least one ID value indicating what type(s) of metadata is present, and optionally also how many bits of (eg each type) metadata present (if metadata exists). High-level structure with displayed values. One type of metadata that may exist is LPSM, and another type of metadata that may exist is media research metadata (eg, Nielsen Media Researcher metadata); A core element for each identified type of metadata (eg, a core header, protection values, and LPSM Payload ID and LPSM Payload Size values, for each identified type of metadata, as noted above) ), including mid-level structures; and each payload for one core element (eg, an LPSM payload if one is identified as present by the core element, and/or another type of payload if one is identified as present by the core element) low-level structure, including metadata payload).

Data values in this three-level structure may be nested. For example, the protection value(s) for an LPSM payload and/or another metadata payload identified by a core element may be determined after each payload identified by the core element (and thus after the core header of the core element). ) may be included. In one example, the core header may identify an LPSM payload and another metadata payload, and payload ID and payload size values for the first payload (eg, LPSM payload) follow the core header. wherein the first payload itself can follow the ID and size values, the payload ID and payload size value for the second payload can follow the first payload, and the second payload itself can follow these IDs and magnitude values, and the guard bits for both payloads (or for both core element values and both payloads) may follow the last payload.

In some embodiments, if the decoder 101 receives an audio bitstream generated according to an embodiment of the present invention having a cryptographic hash, the decoder is configured to parse and retrieve the cryptographic hash from a data block determined from the bitstream, The block contains Loudness Processing State Metadata (LPSM). Validator 102 may use the cryptographic hash to verify the received bitstream and/or associated metadata. For example, the verifier 102 finds the LPSM valid based on a match between the reference cryptographic hash and the cryptographic hash retrieved from the data block, and then disables the processor 103 on the corresponding audio data. and cause the selection stage 104 to pass (unmodified) audio data. Additionally, alternatively, or alternatively, other types of cryptographic techniques may be used in place of cryptographic hash-based methods.

The encoder 100 of FIG. 2 determines (by the decoder 101 ) whether the post/pre-processing unit has performed (in elements 105 , 106 , and 107 ) one type of loudness processing on the audio data to be encoded. may be determined (in response to the extracted LPSM), thus generating (in the generator 106 ) loudness processing state metadata comprising specific parameters used in and/or derived from previously performed loudness processing. In some implementations, encoder 100 can generate processing state metadata representing processing history for audio content (and encoded bits output therefrom) as long as the encoder knows the types of processing being performed on the audio content. can be included in the stream).

3 is a block diagram of a decoder that may be used with system 10 of FIG. 1 . None of the components or elements of decoder 200 and post-processor 300 are hardware, software, or a combination of hardware and software, such as one or more processes and/or one or more circuits (eg, ASICs; FPGAs, or other integrated circuits). The decoder 200 includes a frame buffer 201 , a parser 205 , an audio decoder 202 , an audio state verification stage (verifier) 203 , and a control bit generation stage 204 , connected as shown. do. The decoder 200 may include other processing elements (not shown). The frame buffer 201 (buffer memory) stores (eg, in a non-transitory manner) at least one frame of an encoded audio bitstream received by the decoder 200 . A sequence of frames of the encoded audio bitstream is asserted from the buffer 201 to the parser 205 . parser 205 extracts loudness processing state metadata (LPSM) and other metadata from each frame of encoded input audio, and asserts at least said LPSM to audio state verifier 203 and stage 204; coupled and configured to assert the LSPM (eg, to the post-processor 300 ) as output, extract audio data from the encoded input audio, and assert the extracted audio data to the decoder 202 . The encoded audio bitstream input to the decoder 200 may be one of an AC-3 bitstream, an E-AC-3 bitstream, or a Dolby E bitstream.

The system of FIG. 3 also includes a post-processor 300 . Post-processor 300 includes frame buffer 301 and other processing elements (not shown) including at least one processing element coupled to buffer 301 . The frame buffer 301 stores (eg, in a non-transitory manner) at least one frame of the decoded audio bitstream received from the decoder 200 by the post-processor 300 . The processing elements of the post-processor 300 use the metadata (including LPSM values) output from the decoder 202 and/or the control bits output from the stage 204 of the decoder 200 , the buffer 301 . Combined and configured to receive and adaptively process a sequence of frames of a decoded audio bitstream output from In one embodiment, the post-processor 300 adapts to the decoded audio data using the LPSM values (eg, based on a loudness processing state, indicated by the LPSM, and/or one or more audio data characteristics). and perform enemy loudness processing. Various implementations of decoder 200 and post-processor 300 are configured to perform different embodiments of methods in accordance with the embodiments described herein.

The audio decoder 202 of the decoder 200 decodes the audio data extracted by the parser 205 to generate decoded audio data, and outputs the decoded audio data as an output (eg, the post-processor 300 ) to) is configured to assert. The state verifier 203 is configured to authenticate and verify the asserted LPSM (and optionally other metadata). In some embodiments, the LPSM is (or is included in) a data block included in an input bitstream (eg, according to an embodiment of the present invention). The block is a cryptographic hash (hash-based message) for processing LPSM (and optionally also other metadata) and/or basic audio data (provided from parser 205 and/or decoder 202 to verifier 203 ). authentication code or "HMAC"). The data block may be digitally signed in these embodiments, so that the downstream audio processing unit can relatively easily authenticate and verify the processing state metadata.

Other cryptographic methods including, but not limited to, any of one or more non-HMAC cryptographic methods may be used to ensure secure transmission and reception of LPSM and/or underlying audio data (eg, at verifier 203 ). It can be used for verification. For example, verification (using these cryptographic methods) can determine whether loudness processing state metadata and corresponding audio data included in a bitstream are subject to (and/or result from) certain loudness processing (as indicated by the metadata). ) and in each audio processing receiving a typical audio bitstream of the present invention to determine if it has not changed after performing this particular loudness processing.

The state verifier 203 asserts the control data to the control bit generator 204 and/or as output (eg, to the post-processor 300 ) to indicate the results of the verification operation. assert In response to the control data (and optionally also other metadata extracted from the input bitstream), stage 204 may generate (and assert to post-processor 300 ) one of the following: Control bits indicating whether the decoded audio data output from the decoder 202 has been subjected to a specific type of loudness processing (LPSM indicates that the audio data output from the decoder 202 has been subjected to a specific type of loudness processing; when the control bits from verifier 203 indicate that the LPSM is valid); or control bits indicating that the decoded audio data output from the decoder 202 should be subjected to a specific type of loudness processing (eg, the LPSM indicates that the audio data output from the decoder 202 has not been subjected to a specific type of loudness processing). when indicating a tone, or when the LPSM indicates that the audio data output from the decoder 202 has undergone some type of loudness processing but the control bits from the verifier 203 indicate that the LPSM is invalid).

Alternatively, the decoder 200 asserts the LPSM (and any other metadata) extracted by the decoder 202 from the input bitstream to the post-processor 300, which in turn the LPSM Perform loudness processing on the decoded audio data using LPSM, or perform verification of the LPSM and then perform loudness processing on the decoded audio data using the LPSM if the verification indicates that the LPSM is valid.

In some embodiments, if the decoder 201 receives an audio bitstream generated according to an embodiment of the present invention having a cryptographic hash, the decoder is configured to parse and retrieve the cryptographic hash from a data block determined from the bitstream, The block contains Loudness Processing State Metadata (LPSM). The verifier 203 may use the cryptographic hash to verify the received bitstream and/or associated metadata. For example, if the verifier 203 finds the LPSM valid based on a match between the reference cryptographic hash and the cryptographic hash retrieved from the data block, the downstream audio data of the bitstream (unaltered) is passed through. may signal to a processing unit (eg, post-processor 300 , which may be or include a volume leveling unit). Additionally, alternatively, or alternatively, other types of cryptographic techniques may be used in place of cryptographic hash-based methods.

In some implementations of the decoder 100 , the received (and buffered in memory 201 ) encoded bitstream is an AC-3 bitstream or an E-AC-3 bitstream and includes audio data segments (eg, AB0 to AB5 segments of the frame shown in Figure 4) and metadata segments, wherein the audio data segments represent audio data, each of at least some of the metadata segments comprising loudness processing state metadata (LPSM) ) is included. The decoder stage 202 is configured to extract an LPSM having the following format from the bitstream. Each of the metadata segments containing the LPSM is stored in the "addbsi" field of the bitstream information ("BSI") segment of the frame of the bitstream, or in the ancillary data field (e.g., shown in FIG. 4) at the end of the frame of the bitstream. included in the AUX segment). A frame of a bitstream may contain one or two metadata segments, each containing an LPSM, and if a frame contains two metadata segments, one is in the addbsi field of the frame and the other is in the frame's addbsi field. It exists in the AUX field. Each metadata segment containing an LPSM includes an LPSM payload (or container) segment with the following format: a header (eg, a syncword identifying the start of the LPSM payload, followed by identification values, eg, in the table below) 2, including LPSM format version, length, duration, count, and substream associated values); and after the header, at least one dialog display value indicating whether the corresponding audio data displays a dialog or not (eg, which channels of the corresponding audio data display a dialog) parameter "Dialog Channel(s)" in Table 2); at least one loudness rule compliance value indicating whether the corresponding audio data conforms to the indicated set of loudness rules (eg, the parameter “loudness rule type” in Table 2); At least one loudness processing value indicative of at least one type of loudness processing performed on the corresponding audio data (eg, one or more of the parameters “Dialog Gating Loudness Correction Flag”, “Loudness Correction Type” in Table 2) ; and at least one loudness value indicative of at least one loudness (eg, peak or average loudness) characteristic of the corresponding audio data (eg, the parameters “ITU Relative Gating Loudness”, “ITU Speech Gating Loudness” in Table 2) , "ITU (EBU 3341) short-term 3s loudness", and one or more of "real peaks").

In some implementations, the decoder stage 202 is configured to extract, from an "addbsi" field or ancillary data field of a frame of the bitstream, each metadata segment having the following format: a core header (eg, metadata a syncword identifying the start of a segment followed by at least one identifying value, eg, including core element version, length, and duration, extended element count, and substream associated values indicated in Table 1 below); and after the core header, at least one protection value useful for at least one of decryption, authentication, or verification of at least one of loudness processing state metadata or corresponding audio data (eg, the HMAC digest and audio fingerprint values of Table 1). ); and also after the core header, if the metadata segment contains LPSM, an LPSM Payload Identification (“ID”) and LPSM Payload Size that identifies the following metadata as the LPSM Payload and indicates the size of the LPSM Payload. The LPSM payload (or container) segment (eg, with the specified format) conforms to LPSM Payload ID and LPSM Payload Size values.

More generally, the encoded audio bitstream generated by an embodiment has a structure that provides a mechanism for labeling metadata elements and sub-elements as core (mandatory) or extension (optional elements). This allows the data rate of the bitstream (including its metadata) to scale across multiple applications. The core (mandatory) elements of the bitstream syntax should also be able to signal that extension (optional) elements associated with the audio content are present (in-band) and/or in a remote location (out-of-band).

In some embodiments, the core element(s) are required to be present in every frame of the bitstream. Some sub-elements of the core elements are optional and may be present in any combination. Extended elements are not required to be present every frame (to limit bitrate overhead). Thus, extension elements may be present in some frames and not in others. Some sub-elements of the extension element are optional and may be present in any combination, while some sub-elements of the extension element may be mandatory (ie, if the extension element is present in a frame of the bitstream).

In some embodiments, an encoded audio bitstream comprising an audio data segment and a sequence of metadata segments is generated (eg, by an audio processing unit embodying the present invention). The audio data segments represent audio data, each of at least some of the metadata segments including loudness processing state metadata (LPSM) and the audio data segments are time-division multiplexed with the metadata segments. In some embodiments in this class, each of the metadata segments has a format as will be described herein. In one format, the encoded bitstream is an AC-3 bitstream or an E-AC-3 bitstream, and each of the metadata segments comprising an LPSM is a bitstream information (“BSI”) segment of a frame of the bitstream. It is included as additional bit stream information in the "addbsi" field (shown in FIG. 6 ), or in the auxiliary data field of a frame of the bitstream (eg, by stage 107 of encoder 100 ). Each of the frames includes a core element in the addbsi field of the frame having the format shown in Table 1 of FIG. 8 .

In one format, each of the addbsi (or auxiliary data) fields comprising the LPSM is a core header (and optionally also additional core elements), and after the core header (or core header and other core elements), the next contains the LPSM values (parameters) of: a payload ID (identifying the metadata as an LPSM) that conforms to the core element values (eg, as specified in Table 1); payload size (indicating the size of the LPSM payload) followed by the payload ID; and LPSM data (according to payload ID and payload size values) having a format as indicated in Table 2 of FIG. 9 .

In a second format of the encoded bitstream, the bitstream is an AC-3 bitstream or an E-AC-3 bitstream, and each of the metadata segments comprising the LPSM (eg, stage 107 of encoder 100 ) )) in one of the following: the "addbsi" field of the bitstream information ("BSI") segment of the frame of the bitstream (shown in FIG. 6 ); or an auxiliary data field at the end of the frame of the bitstream (eg, the AUX segment shown in FIG. 4 ). A frame may contain one or two metadata segments, each containing an LPSM, if the frame contains two metadata segments, one in the addbsi field of the frame and the other in the AUX field of the frame. do. Each metadata segment including the LPSM has the format specified above with reference to Tables 1 and 2 above (ie it contains the core elements specified in Table 1, followed by the specified Payload ID (Metadata as LPSM). ) and payload size values, followed by the payload (LPSM data in the format as shown in Table 2).

In another, the encoded bitstream is a Dolby E bitstream, and each of the metadata segments containing the LPSM is the first N sample positions of the Dolby E guard band interval. A Dolby E bitstream containing this metadata segment comprising LPSM contains, for example, a value indicating the LPSM payload length signaled in the Pd word of the SMPTE 337M preamble (SMPTE 337M Pa word repetition rate is the associated video frame rate). can remain the same as ).

In a format in which the encoded bitstream is an E-AC-3 bitstream, each of the metadata segments comprising the LPSM (eg, by stage 107 of the encoder 100 ) contains bitstream information of a frame of the bitstream. ("BSI") Included as additional bitstream information in the "addbsi" field of the segment. Additional aspects of encoding an E-AC-3 bitstream with LPSM in this format are described as follows: (1) During generation of the E-AC-3 bitstream (inserting LPSM values into the bitstream) While the E-AC-3 encoder is "active" for every generated frame (syncframe), the bitstream must contain a block of metadata (including LPSM) carried in the addbsi field of the frame. The bits required to carry a metadata block should not increase the encoder bitrate (frame length); (2) Every metadata block (including LPSM) shall contain the following information: Loudness_Correction_Type_Flag: where '1' indicates that the loudness of the corresponding audio data has been calibrated upstream from the encoder. and '0' indicates that the loudness has been corrected by a loudness corrector built into the encoder (eg, the loudness processor 103 of the encoder 100 of FIG. 2); Speech_Channel: Indicates which source channel(s) contains speech (over the previous 0.5 seconds). If no speech was detected, it will be marked like this; Speech_Loudness: indicates the integrated speech loudness of each corresponding audio channel containing speech (over the previous 0.5 seconds); ITU_Loudness: indicates the integrated ITU BS.1770-2 loudness of each corresponding audio channel; Gain: Loudness composite gain(s) for reversal at the decoder (to show reversibility).

While the E-AC-3 encoder (inserting LPSM values into the bitstream) is "active" and receives an AC-3 frame with a "trust" flag, its loudness controller (eg, encoder 100 in FIG. 2 ) ) of the loudness processor 103) is bypassed. The “trusted” source Dialnorm and DRC values are passed (eg, by the generator 106 of the encoder 100 ) to the E-AC-3 encoder component (eg, the stage 107 of the encoder 100 ). do. LPSM block generation continues and the loudness_correction_type_flag is set to '1'. The loudness controller bypass sequence is synchronized to the beginning of the decoded AC-3 frame where the 'trusted' flag appears. The loudness controller bypass sequence is implemented as follows: the leveler_amount control is reduced from a value of 9 to a value of 0 over 10 audio block periods (ie 53.3 milliseconds) and the leveler_back_end_meter control is It is placed in bypass mode (this action should be a seamless transition). The term leveler's 'trusted' bypass means that the Dialnorm value of the source bitstream is also re-used at the output of the encoder. (For example, if the 'trusted' source bitstream has a Dialnorm value of -30, the output of the encoder must use -30 for the outbound Dialnorm value).

While the E-AC-3 encoder (which inserts the LPSM values into the bitstream) is "active" and receives an AC-3 frame without a 'trusted' flag, the loudness controller built into the encoder (e.g. the encoder of Figure 2) The loudness processor 103 of 100 is active. LPSM block generation continues and the loudness_correction_type_flag is set to '0'. The loudness controller activation sequence is synchronized to the beginning of the decoded AC-3 frame where the 'trust' flag disappears. The loudness controller activation sequence is implemented as follows: Leveler_Amount control is incremented from a value of 0 to a value of 9 over a period of 1 audio block (ie 5.3 milliseconds) and Leveler_Back_End_Meter control is 'active' placed in ' mode (this operation becomes a seamless transition and includes a back_end-meter integrated reset); and during encoding, the graphical user interface (GUI) displays the following parameters to the user: "Input audio program: [trusted/untrusted]" - the state of this parameter is the presence of a "trusted" flag in the input signal based on; and “Real-Time Loudness Calibration: [Enable/Disable]”—The state of this parameter is based on whether or not this loudness controller built into the encoder is active.

When decoding an AC-3 or E-AC-3 bitstream with LPSM (of the format described) included in the "addbsi" field of the bitstream information ("BSI") segment of each frame of the bitstream, the decoder parses the LPSM block data (in the addbsi field) and delivers the extracted LPSM values to the graphical user interface (GUI). The set of extracted LPSM values is refreshed every frame.

In another format, the encoded bitstream is an AC-3 bitstream or an E-AC-3 bitstream, wherein each of the metadata segments comprising the LPSM is a bitstream information (“BSI”) segment of a frame of the bitstream. Included as additional bit stream information in the "addbsi" field (shown in FIG. 6 ) (or in the Aux segment) (eg, by stage 107 of encoder 100 ). In this format (which is a variation on the format described above with reference to Tables 1 and 2), each of the addbsi (or Aux) fields containing the LPSM contains the following LPSM values: Core Element Specified in Table 1 , followed by the payload ID and payload size values (identifying the metadata as LPSM), followed by a payload (LPSM data) with the following format (similar to the elements indicated in Table 2 above): the version of the LPSM payload. : 2-bit field indicating the version of the LPSM payload; dialchan: A 3-bit field indicating whether the left, right and/or center channels of the corresponding audio data contain a voice dialog. The bit assignment of the dialchan field may be as follows: bit 0 indicating the presence of a dialog in the left channel is stored in the most significant bit of the dialchan field; Bit 2, indicating the presence of a dialog in the central channel, is stored in the least significant bit of the dialchan field. Each bit of the dialchan field is set to '1' if the corresponding channel contains voice dialogue during the previous 0.5 seconds of the program; loudregtyp: 3-bit field indicating which loudness regulation standard the program loudness conforms to. Setting the "loudregtyp" field to '000' indicates that the LPSM does not indicate loudness compliance. For example, one value of this field (eg, 000) may indicate that compliance with a loudness regulation standard is not indicated, and another value of this field (eg, 001) may indicate that the program's audio data It may indicate conformance to the ATSC A/85 standard, and another value in this field (eg, 010) may indicate that the audio data of the program conforms to the EBU R128 standard. In the example above, if the field is set to any value other than '000', then the loudcorrdialgat and loudcorrtyp fields in the payload must follow; loudcorrdialgat: 1-bit field indicating whether dialog-gated loudness correction has been applied. If the loudness of the program has been calibrated using dialog gating, the value of the loudcorrdialgat field is set to '1'; otherwise it is set to '0'; loudcorrtyp: 1-bit field indicating the type of loudness correction applied to the program. If the loudness of the program has been calibrated with an infinite look-ahead (file-based) loudness calibration process, the value of the loudcorrtyp field is set to '0'. If the loudness of the program has been calibrated using a combination of real-time loudness measurement and dynamic range control, the value of this field is set to '1'; loudrelgate: 1-bit field indicating whether relative gating loudness data (ITU) is present. If the loudrelgate field is set to '1', a 7-bit ituloudrelgat field shall follow in the payload; loudrelgat: 7-bit field indicating the relative gating program loudness (ITU). This field indicates the integrated loudness of the audio program, measured according to ITU-R BS.1770-2, without any gain adjustments due to the applied Dialnorm and dynamic range compression. Values from 0 to 127 are interpreted as 0.5 LKFS steps, -58 LKFS to +5.5 LKFS; loudspchgate: 1-bit field indicating whether speech-gating loudness data (ITU) is present. If the loudspchgate field is set to '1', a 7-bit loudspchgat field shall follow in the payload; loudspchgat: 7-bit field indicating speech-gating program loudness. This field indicates the integrated loudness of the entire corresponding audio program, measured according to equation (2) of ITU-R BS.1770-3 and without any gain adjustments due to the applied Dialnorm and dynamic range compression. Values from 0 to 127 are 0.5 LKFS steps, interpreted as -58 to +5.5 LKFS; loudstrm3se: 1-bit field indicating whether short-term (3 sec) loudness data is present. If the field is set to '1', a 7-bit loudstrm3s field shall follow in the payload; loudstrm3s: 7-bit field indicating the ungated loudness of the previous 3 seconds of the corresponding audio program, measured according to ITU-R BS.1771-1, without any gain adjustments due to Dialnorm and dynamic range compression applied. Values from 0 to 256 are interpreted as -116 LKFS to +11.5 LKFS with steps of 0.5 LKFS; truepke: 1-bit field indicating whether actual peak loudness data is present. If the truepke field is set to '1', an 8-bit truepk field MUST be followed in the payload; and truepk: an 8-bit field indicating the actual peak sample value of the program, measured according to Annex 2 of ITU-R BS.1770-3 without any gain adjustments due to Dialnorm and dynamic range compression applied. Values from 0 to 256 are interpreted as -116 LKFS to +11.5 LKFS with 0.5 LKFS steps.

In some embodiments, the core element of the metadata segment in the ancillary data field (or "addbsi" field) of the frame of the AC-3 bitstream or E-AC-3 bitstream is a core header (optionally identifying values, e.g. including the core element version), and after the core header: values indicating whether the fingerprint data is included for the metadata of the metadata segment (or other protection values are included), (in the metadata of the metadata segment) Values indicating whether external data (related to the corresponding audio data) exists, a pay for each type of metadata identified by the core element (eg, LPSM, and/or metadata of a type other than LPSM) load ID and payload size values, and protection values for at least one type of metadata identified by the core element. The metadata payload(s) of the metadata segment follow the core header and are (in some cases) nested within the values of the core element.

Optimized loudness and dynamic range system

The secure metadata coding and transmission technique described above is for use with a scalable and extensible system for optimizing loudness and dynamic range across different playback devices, applications, and listening environments, as illustrated in FIG. 1 . do. In an embodiment, the system 10 is configured to normalize the dynamic range and loudness levels of the input audio 11 across various devices that require different target loudness values and have different dynamic range capabilities. To normalize loudness levels and dynamic range, system 10 includes different device profiles with audio content and normalization is done based on these profiles. The profiles may be included by one of the audio processing units in the audio processing chains and the included profiles will be used by a downstream processing unit in the audio processing chain to determine a desired target loudness and dynamic range for a target device. can Additional processing components include device profile management, gain control and wideband (including but not limited to parameters of null band range, actual peak threshold, loudness range, fast/slow time constants (coefficients) and max boost). and/or provide or process information for multiband gain generation functions.

10 illustrates a more detailed diagram of the system of FIG. 1 for a system that provides optimized loudness and dynamic range control, under some embodiments. For system 321 of FIG. 10 , the encoder stage includes a core encoder component 304 that encodes audio input 303 into a digital format suitable for transmission to decoder 312 . The audio is processed so that it can be reproduced in a variety of different listening environments, each of which may require different loudness and/or dynamic range target settings. Thus, as shown in FIG. 10 , the decoder is a digital-to-analog converter for playback through a variety of different driver types, including full range speakers 320 , small speakers 322 , and headphones 324 . 316 outputs a digital signal that is converted to an analog format. These drivers illustrate only a few examples of possible regenerative drivers, and any transducer or driver of any suitable size and type may be used. Further, the drivers/transducers 320 - 324 of FIG. 10 may be implemented in any suitable playback device for use in any corresponding listening environment. Device types may include, for example, AVRs, televisions, stereo equipment, computers, mobile phones, tablet computers, MP3 players, etc., and listening environments are, for example, auditoriums, home fields, cars, listening booths, and the like.

Because the range of playback environments and driver types can vary from very small private contexts to very large public places, the breadth of possible and optimal playback loudness and dynamic range configurations can vary greatly depending on the content type, background noise levels, etc. may vary. For example, in a home theater environment, wide dynamic range content may be played through surround sound equipment and narrower dynamic range content may be played through conventional television systems (such as flat panel LED/LCD types), whereas The very narrow dynamic range mode can be used for certain listening conditions where large level variations are undesirable (eg at night or on devices with severe acoustic output power limitations, eg mobile phone/tablet internal speakers or headphone output). ). In portable or mobile listening contexts, such as using a small computer or dock speakers, or headphones/earbuds, the optimal dynamic range of playback may vary depending on the environment. For example, in a quiet environment, the optimal dynamic range may be greater compared to a noisy environment. Embodiments of the adaptive audio processing system of FIG. 10 will change the dynamic range to make the audio content easier to understand depending on parameters, such as the listening device environment and playback device type.

11 is a table illustrating different dynamic range requirements for various playback devices and background listening environments in an exemplary use case. Similar requirements can be obtained for loudness. Different dynamic range and loudness requirements create different profiles used by the optimization system 321 . System 321 includes a loudness and dynamic range measurement component 302 that analyzes and measures the loudness and dynamic range of input audio. In an embodiment, the system analyzes the entire program content to determine an overall loudness parameter. In this context, loudness refers to the long term program loudness or average loudness of a program, where a program is a single unit of audio content, such as a movie, television show, advertisement, or similar program content. Loudness is used to provide an indication of the artistic dynamic range profile used by content creators to control how the audio will be played. Loudness is related to Dialnorm metadata values in that Dialnorm represents the average dialog loudness of a single program (eg, movie, show, advertisement, etc.). Short-term dynamic range quantifies variations in signals over a much shorter period of time than program loudness. For example, short-term dynamic range may be measured in the order of seconds, whereas program loudness may be measured over a span of minutes or even hours. Short-term dynamic range provides a program loudness-independent protection mechanism to ensure that overloading does not occur for various playback profiles and device types. In an embodiment, the loudness (long term program loudness) target is based on dialog loudness and the short term dynamic range is based on relative-gated and/or non-gated loudness. In this case, the specific DRC and loudness components in the system are context-aware with respect to content type and/or target device types and characteristics. As part of this context-aware capability, the system determines whether a device is a member of specific groups of devices that are optimized for specific DRC and loudness playback conditions, such as AVR type devices, televisions, computers, portable devices, and the like. and analyze one or more characteristics of the output device to determine.

A pre-processing component analyzes the program content to determine loudness, peaks, actual peaks, and quiet periods to generate unique metadata for each profile of a plurality of different profiles. In embodiments, the loudness may be dialog-gated loudness and/or relative-gated loudness. Different profiles define various DRC (dynamic range control) and target loudness modes, where different gain values are generated at the encoder depending on the characteristics of the source audio content, the desired target loudness and playback device type and/or environment. The decoder can provide different DRC and target loudness modes (enabled by the profiles mentioned above) and off/disabled DRC and Target Loudness, on home theater systems with loudness normalization with a target of -31 LKFS and providing adequate dynamic range compression via gain values generated at the encoder (specifically for this playback mode and/or device profile). Off/disabled DRC and loudness normalization with target of -31 LKFS line mode for playback; RF mode for playback through TV speakers providing a large amount of dynamic range compression with loudness normalization with a target of either -24, -23 or -20 LKFS, compression with loudness normalization at a target of -14 LKFS , an intermediate mode for playback across computers or similar devices, and a portable mode that provides a very large dynamic range compression with a loudness normalization target of -11 LKFS. Target loudness values of -31, -23/-20, -14, and -11 LKFS are intended to be examples of different playback/device profiles that may be defined for the system under some embodiments, and any other suitable target Loudness values may be used and the system generates appropriate gain values specific to these playback modes and/or device profile. Moreover, the system is scalable and adaptable so that different playback devices and listening environments can be accommodated and loaded into the encoder by defining a new profile at the encoder or otherwise. In this way, new and unique playback/device profiles can be created to support improved or different playback devices for further applications.

In an embodiment, the gain values are determined by any suitable processing component of system 321 , such as encoder 304 , decoder 312 , or transcoder 308 , or any associated pre-processing component associated with the encoder. or in any post-processing component associated with the decoder.

13 is a block diagram illustrating the interface between different profiles for various different playback device classes, under an embodiment. As shown in FIG. 13 , the encoder 502 receives an audio input 501 and one of several different possible profiles 506 . The encoder combines the audio data with the selected profile to produce an output bitstream file that is present at or associated with the target playback device and processed at a decoder component. In the example of FIG. 13 , the different playback devices may be a computer 510 , a mobile phone 512 , an AVR 514 , and a television 516 , although many other output devices are also possible. Each of devices 510 - 516 includes or is coupled to speakers (including drivers and/or transducers), such as drivers 320 - 324 . The combination of processing, output ratings, and sizes of playback devices and associated speakers generally dictates which profile is most optimal for that particular target. Thus, the profiles 506 may be specifically defined for playback via AVRs, TVs, mobile speakers, mobile headphones, and the like. They may also be defined for specific operating modes or states, such as quiet mode, night mode, outdoor, indoor, and the like. The profiles shown in FIG. 13 are merely exemplary modes and any suitable profile may be defined, including custom profiles for specific targets and environments.

13 illustrates an embodiment in which the encoder 502 receives the profiles 506 and generates appropriate parameters for loudness and DRC processing, but the parameters generated based on the profile and audio content are the encoder, decoder, transcoder , it should be noted that it may be performed on any suitable audio processing unit, such as a pre-processor, post-processor, etc. For example, each output device 510-516 of FIG. 13 may have a file 504 sent from the encoder 502 to enable an adaptation of loudness and dynamic range to match the device type of the device or target output device. ) has or is coupled to a decoder component that processes metadata in the bitstream.

In an embodiment, the dynamic range and loudness of the audio content is optimized for each possible playback device. This is achieved by controlling the short-term dynamic range while targeting long-term loudness to optimize the audio experience for each of the target playback modes (by controlling the signal dynamics, sample peaks and/or actual peaks). Different metadata elements are defined for long term loudness and short term dynamic range. As shown in Figure 10, component 302 is configured to derive the relevant properties for both of these distinct DR components, the entire input audio signal (or portions thereof, such as the speech component, if applicable). ) is analyzed. This allows different gain values to be defined for artistic gains versus clip (overload protection) gain values.

These gain values for long term loudness and short term dynamic range are then mapped to profile 305 to generate parameters that describe loudness and dynamic range control gain values. These parameters are combined with the encoded audio signal from the encoder 304 in a multiplexer 306 , or similar component for generation of a bitstream that is transmitted to the decoder stage via a transcoder 308 . The bitstream input to the decoder stage is demultiplexed in the demultiplexer 310 . It is then decoded in decoder 312 . The gain component 314 then corresponds to the appropriate profile to generate digital audio data that is processed via the DACS unit 416 for playback via the appropriate playback devices and drivers or transducers 320 - 324 . apply the benefits of

14 is a table illustrating the correlation between long term loudness and short term dynamic range for a plurality of defined profiles, under an embodiment. As shown in Table 4 of FIG. 14 , each profile includes a set of gain values that dictate the amount of dynamic range compression (DRC) applied at each of the target devices or at the decoder of the system. Each of the N profiles, denoted as profiles 1 through N, sets certain long term loudness parameters (eg Dialnorm) and overload compression parameters by dictating the corresponding gain values applied in the decoder stage. The DRC gain values for the profiles may be defined by an external source accepted by the encoder, or they may be generated internally within the encoder as default gain values if no external values are provided.

In an embodiment, the gain values for each profile are the time constants needed to effect fast/slow attack and fast/slow release of the final DRC gains for each possible device profile and/or target loudness as well as the time constants required to perform fast/slow release. certain characteristics of the audio signal, such as peak, actual peak, short-term loudness of dialog or overall short-term loudness, or a combination of both (hybrid), to calculate a static gain based on a selected profile (i.e., transmission characteristic or curve). It is implemented in the DRC gain words calculated based on the analysis. As described above, these profiles may exist at the encoder, decoder, or are generated externally and passed from the content producer to the encoder via external metadata.

In an embodiment, the gain values may be a wideband gain that applies the same gain across all frequencies of the audio content. Alternatively, the gain may consist of multi-band gain values such that different gain values are applied to different frequencies or frequency bands of the audio content. In the multi-channel case, each profile may constitute a matrix of gain values indicating gains for different frequency bands instead of a single gain value.

Referring to FIG. 10 , in an embodiment information regarding properties or characteristics of the capabilities and configurations of the listening environments and/or playback devices is provided by the decoder stage to the encoder stage by way of a feedback link 330 . . Profile information 332 is also input to encoder 304 . In an embodiment, the decoder analyzes the metadata in the bitstream to determine whether a loudness parameter for the first group of audio reproduction devices is available in the bitstream. If so, it sends parameters downstream for use in rendering the audio. Otherwise, the encoder analyzes certain characteristics of the devices to derive parameters. These parameters are then sent to the downstream rendering component for playback. The encoder also determines an output device (or group of output devices comprising an output device) to render the received audio stream. For example, the output device may be determined to be a cell phone or to belong to a group of portable devices. In an embodiment, the decoder uses the feedback link 330 to indicate to the encoder the determined output device or group of output devices. For this feedback, a module connected to the output device (eg, a module in a soundcard connected to headsets or connected to speakers of a laptop) is the identity of the output device or the identity of the group of devices containing the output device. can be displayed to the decoder. The decoder transmits this information to the encoder via a feedback link 330 . In an embodiment, the decoder performs the decoder to determine loudness and DRC parameters. In an embodiment, the decoder determines loudness and DRC parameters. In this embodiment, instead of transmitting information over feedback link 330 , the decoder uses information about the determined device or group of output devices to determine loudness and DRC parameters. In another embodiment, another audio processing unit determines loudness and DRC parameters and the decoder sends the information to the audio processing unit instead of the decoder.

12 is a block diagram of a dynamic range optimization system, under an embodiment; As shown in FIG. 12 , the encoder 402 receives the input audio 401 . The encoded audio is combined in a multiplexer 409 with parameters 404 generated from a selected compression curve 422 and a Dialnorm value 424 . The resulting bitstream is sent to a demultiplexer 411 that generates audio signals that are decoded by a decoder 406 . The parameters and Dialnorm values are used by the gain calculation unit 408 to generate gain levels that drive the amplifier 410 for amplification of the decoder output. 12 illustrates how dynamic range control is parameterized and inserted into the bitstream. Loudness can also be parameterized and inserted into the bitstream using similar components. In an embodiment, an output reference level control (not shown) may also be provided to the decoder. Although the figure illustrates loudness and dynamic range parameters as determined and inserted at the encoder, similar determinations may be made in other audio processing units such as pre-processor, decoder, and post-processor.

15 illustrates examples of loudness profiles for different types of audio content, under an embodiment. As shown in FIG. 15 , exemplary curves 600 and 602 represent the input loudness (in LKFS) versus gain centered around zero LKFS. Different types of content show different curves as shown in FIG. 15 , where curve 600 represents speech and curve 602 may represent standard film content. As shown in Figure 15, speech content is subject to a greater amount of gain compared to film content. 15 is intended to be examples of representative profile curves for certain types of audio content, other profile curves may also be used. As shown in FIG. 15 , certain aspects of the profile characteristics are used to derive relevant parameters for the optimization system. In an embodiment, these parameters include: null bandwidth, cut ratio, boost ratio, maximum boost, FS attack, FS attenuation, holdoff, peak limit, and target level loudness. Other parameters may be used in addition to or alternatively to at least some of these parameters depending on application requirements and system constraints.

16 is a flow diagram illustrating a method of optimizing loudness and dynamic range across playback devices and applications, under an embodiment. Although the figure illustrates loudness and dynamic range optimization as being performed at the encoder, similar optimizations may be performed in other audio processing units such as pre-processor, decoder, and post-processor. As shown in process 620, the method begins with an encoder stage receiving an input signal from a source (603). The encoder or pre-processing component then determines whether the source signal has been subjected to a process to achieve target loudness and/or dynamic range (604). The target loudness corresponds to a long-term loudness and may be defined externally or internally. If the source signal has not been subjected to processing to achieve the target loudness and/or dynamic range, the system performs 608 appropriate loudness and/or dynamic range control operations; Otherwise, if the source signal has been subjected to such a loudness and/or dynamic range control operation, the system is configured to skip the loudness control and/or dynamic range operations to allow the original process to dictate an appropriate long term loudness and/or dynamic range operation. Enter bypass mode (606). Appropriate gain values for either bypass mode 606 or performance mode 608 (which may be single wideband gain values or frequency-dependent multi-band gain values) are then applied 612 to the decoder.

bitstream format

As previously described, a system for optimizing loudness and dynamic range ensures that the metadata and audio content transmitted in the bitstream between the encoder and the decoder, or between the source and rendering/playback devices, are not separated from each other or or other networks or other proprietary equipment, such as service provider interfaces, to ensure that it has not been corrupted during transmission, etc., using a secure, extensible metadata format. This bitstream provides a mechanism for signaling the encoder and/or decoder components to adapt the loudness and dynamic range of the audio signal to match the audio content and output device characteristics via appropriate profile information. In an embodiment, the system is configured to determine a low bit rate encoded bitstream to be transmitted between an encoder and a decoder, wherein the encoded loudness information via metadata includes characteristics for one or more output profiles. A description of a bitstream format for use with a loudness and dynamic range optimization system follows under an embodiment.

An AC-3 encoded bitstream contains 1 to 6 channels and metadata for audio content. The audio content is audio data compressed using perceptual audio coding. The metadata includes several audio metadata parameters intended for use in modifying the sound of a program delivered to the listening environment. Each frame of the AC-3 encoded audio bitstream contains audio content and metadata for 1536 samples of digital audio. For a sampling rate of 48 kHz, this represents a rate of 32 milliseconds of digital audio or 31.25 frames per second of audio.

Each frame of an E-AC-3 encoded audio bitstream contains 256, 512, 768, or 1536 samples of digital audio, depending on whether the frame contains 1, 2, 3, or 6 blocks of audio data, respectively. It contains audio content and metadata about them. For a sampling rate of 48 kHz, this represents a rate of 5.333, 10.667, 16 or 32 milliseconds of digital audio, respectively, or 189.9, 93.75, 62.5 or 31.25 frames per second of audio, respectively.

As indicated in FIG. 4, each AC-3 frame is divided into sections (segments) containing: a synchronization word (SW) and the first of two error correction words (CRC1) (in FIG. as shown) a synchronization information (SI) section comprising; a bitstream information (BSI) section containing most of the metadata; 6 audio blocks AB0 to AB5 containing data compressed audio content (and may also contain metadata); waste bits (W) including any unused bits left after the audio content has been compressed; Auxiliary (AUX) information section, which may contain more metadata; and the second of the two error correction words (CRC2).

As shown in Fig. 7, each E-AC-3 frame is divided into sections (segments) containing: Synchronization information (SI) containing a synchronization word (SW) (as shown in Fig. 5) ) section; a bitstream information (BSI) section containing most of the metadata; audio blocks AB0 to AB5 between 1 and 6 containing data compressed audio content (and may also contain metadata); waste bits (W) including any unused bits left after the audio content has been compressed; Auxiliary (AUX) information section, which may contain more metadata; and Error Correction Word (CRC).

In the AC-3 (or E-AC-3) bitstream, there are several audio metadata parameters that are specifically intended for use in modifying the sound of a program delivered to the listening environment. One of the metadata parameters is the Dialnorm parameter, which is included in the BSI segment.

As shown in Fig. 6, the BSI segment of the AC-3 frame includes a 5-bit parameter ("Dialnorm") indicating the Dialnorm value for the program. The 5-bit parameter ("Dialnorm2") indicating the Dialnorm value for the second audio program carried in the same AC-3 frame indicates that the audio coding mode ("acmod") of the AC-3 frame is either double-mono or "1". +1" is included if "0", indicating that the channel configuration is in use.

The BSI segment also includes an “addbsie” bit followed by a flag (“addbsie”) indicating the presence (or absence) of additional bit stream information, a “addbsil” value followed by a parameter indicating the length of any additional bit stream information ("addbsil"), and up to 64 bits of additional bit stream information ("addbsi") following the "addbsil" value. The BSI segment may include other metadata values not specifically shown in FIG. 6 .

Aspects of one or more embodiments described herein may be implemented in an audio system that processes audio signals for transmission over a network that includes one or more computers or processing devices executing software instructions. Any of the described embodiments can be used alone or with each other in any combination. While various embodiments may have been motivated by various deficiencies of the prior art and may be discussed or suggested in one or more portions herein, the embodiments do not necessarily address any of these deficiencies. In other words, different embodiments may address different drawbacks that may be discussed herein. Some embodiments may only partially address some or only one drawback that may be discussed herein, and some embodiments may not address any of these drawbacks.

Aspects of the systems described herein may be implemented in a suitable computer-based sound processing network environment for processing digital or digitized audio files. Portions of the adaptive audio system may include one or more networks comprising any desired number of individual machines, including one or more routers (not shown) that act to buffer and route transmitted data among the computers. can Such a network may be established over a variety of different network protocols, and may be the Internet, a wide area network (WAN), a local area network (LAN), or any combination thereof.

One or more of the components, blocks, processes, or other functional components may be implemented via a computer program that controls execution of a processor-based computing device of the system. The various functions disclosed herein may be implemented using any number of combinations of hardware, firmware and/or various machine-readable or computer-readable with respect to their behavior, register transfers, logical components, and/or other characteristics. It should also be noted that may be described as embodied data and/or instructions in possible media. Computer-readable media in which such formatted data and/or instructions may be embodied include, but are not limited to, physical (non-transitory), non-volatile, Includes storage media.

Unless the context clearly requires otherwise, throughout the description and claims, the words "comprises", "comprising" and the like are in an inclusive sense, i.e., not limited thereto, as opposed to an exclusive or exhaustive sense. , which will be construed as “including”. Words using the singular or plural number also include the plural or singular number, respectively. Additionally, the words "herein," "below," "above," "below," and words of similar meaning refer to the present application as a whole and not to any specific portions of the present application. or "when used in reference to a list of two or more items, the word covers all of the following interpretations of the word: any of the items in the list. all of the items in the list and in the list. any combination of items in

While one or more implementations have been described by way of example and with respect to specific embodiments, it will be understood that one or more implementations are not limited to the disclosed embodiments. On the contrary, it is intended to cover various modifications and similar arrangements, as will be apparent to those skilled in the art. Therefore, the scope of the appended claims is to be accorded the broadest interpretation so as to cover all such modifications and similar scopes.

10: System 11: Input
12: pre-processing unit 14: encoder
16: signal analysis and metadata correction unit 18: transcoder
20: decoder 24: post-processing unit
100: encoder 101: decoder
102: audio state verifier 103: loudness processing stage
104: audio stream selection stage 105: encoder
106: metadata generator 107: stuffer/formatter stage
108: dialog loudness measurement subsystem 109: frame buffer
110: frame buffer 111: parser
150: subsystem 152: decoder
200: decoder 201: frame buffer
202: audio decoder 203: audio state verifier
204: control bit generation stage 205: parser
300: post-processor 301: frame buffer
303: audio input 304: core encoder component
306: multiplexer 308: transcoder
310: demultiplexer 312: decoder
314: gain component 316: digital-to-analog converter
320: full range speaker 322: small speaker
324: headphones 401: input audio
402: encoder 406: decoder
408: gain calculation unit 409: multiplexer
410: amplifier 411: demultiplexer
416: DACS unit 501: audio input
502: encoder 506: profile
510: computer 512: mobile phone
514: AVR 516: Television

Claims (8)

An audio processing device for decoding one or more frames of an encoded audio bitstream, the encoded audio bitstream comprising audio data and metadata for a plurality of dynamic range control (DRC) profiles. In:
a bitstream parser configured to parse the encoded audio bitstream and extract encoded audio data and metadata for one or more of the DRC profiles; and
an audio decoder configured to decode the encoded audio data and apply a DRC gain to the decoded audio data;
Each DRC profile is suitable for at least one device type or listening environment;
the audio decoder selects one or more DRC profiles in response to information about the audio processing device or a listening environment;
a DRC gain applied to the decoded audio data corresponds to one or more selected DRC profiles;
wherein the DRC gain corresponding to the one or more selected DRC profiles is determined from DRC parameters for the one or more selected DRC profiles included in metadata of the encoded audio bitstream. The method of claim 1,
and the DRC parameter represents a static gain transmission characteristic and a gain smoothing time constant. 3. The method of claim 2,
wherein the time constants include slow and fast attack time constants and slow and fast release time constants. 3. The method of claim 2,
wherein the static gain transfer characteristic comprises a null band range and a maximum boost. An audio processing method for decoding one or more frames of an encoded audio bitstream, performed by an audio processing device, wherein the encoded audio bitstream comprises audio data and metadata for a plurality of dynamic range control (DRC) profiles. In the audio processing method comprising:
parsing the encoded audio bitstream and extracting encoded audio data and metadata for one or more of the DRC profiles; and
decoding the encoded audio data and applying a DRC gain to the decoded audio data;
each DRC profile is suitable for at least one device type or listening environment;
one or more DRC profiles are selected in response to information about the audio processing device or a listening environment;
a DRC gain applied to the decoded audio data corresponds to one or more selected DRC profiles;
wherein the DRC gain corresponding to the one or more selected DRC profiles is determined from DRC parameters for the one or more selected DRC profiles included in metadata of the encoded audio bitstream. A non-transitory computer-readable recording medium comprising a series of instructions that, when executed by an audio decoding apparatus, cause the audio decoding apparatus to perform the method of claim 2 . A software program suitable for execution on a processor comprising a series of instructions that, when executed by an audio decoding apparatus, cause the method of claim 5 to be performed. A computer program product comprising a series of instructions that, when executed on a computer, perform the method of claim 5 .

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